A software-based conferencing platform is provided. The platform comprises a plurality of audio sources providing input audio signals, the audio sources including a virtual audio device driver configured to receive far-end input audio signals from a conferencing software module, and a network audio library configured to receive near-end input audio signals from one or more near-end audio devices. The platform further comprises a digital signal processing component configured to receive the input audio signals from the audio sources and generate audio output signals based the received signals, the digital signal processing component comprising an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end input audio signals.
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1. A software-based conferencing platform comprising: a plurality of audio sources providing input audio signals, the audio sources including: a virtual audio device driver configured to receive far-end input audio signals from a conferencing software module, and a network audio library configured to receive near-end input audio signals from one or more near-end audio devices; and a digital signal processing component configured to receive the input audio signals from the audio sources and generate audio output signals based on the received signals, the digital signal processing component comprising an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end input audio signals, wherein each of the virtual audio device driver, the network audio library, and the digital signal processing component is stored on a local computing device connected to said one or more near-end audio devices and comprising said conferencing software module, and wherein the virtual audio device driver appears as a hardware audio device to the conferencing software module.
2. The platform of claim 1 , wherein the digital signal processing component further comprises an automixing module configured to mix two or more of the near-end input audio signals to generate an automix output signal.
This invention relates to a digital signal processing platform for audio systems, particularly for managing multiple near-end input audio signals in real-time communication applications. The problem addressed is the need to efficiently process and combine multiple audio inputs from different sources, such as microphones in a conference system, to produce a clear and coherent output signal while minimizing interference and noise. The platform includes a digital signal processing component that processes near-end input audio signals, which are audio signals captured from multiple sources in the same environment. The system enhances these signals by applying techniques such as noise reduction, echo cancellation, and beamforming to improve audio quality. A key feature is an automixing module within the digital signal processing component. This module automatically mixes two or more of the near-end input audio signals to generate an automix output signal. The automixing process dynamically adjusts the contribution of each input signal based on factors such as signal strength, directionality, and noise levels, ensuring that the output signal is balanced and free from excessive overlap or distortion. This allows for seamless integration of multiple audio sources without manual intervention, improving usability in applications like video conferencing, live broadcasting, and multi-microphone setups. The system optimizes audio clarity and intelligibility while reducing computational overhead.
3. The platform of claim 1 , wherein the digital signal processing component further comprises a matrix mixing module configured to generate the audio output signals.
This invention relates to a digital signal processing platform designed to enhance audio signal processing, particularly in systems requiring precise control over multiple audio channels. The platform addresses the challenge of efficiently managing and mixing audio signals in real-time applications, such as live sound reinforcement, broadcasting, or audio production, where multiple input signals must be combined and processed to produce high-quality output signals. The platform includes a digital signal processing component that processes input audio signals to generate output audio signals. A key feature is a matrix mixing module within this component, which dynamically combines and routes audio signals from multiple input channels to multiple output channels. The matrix mixing module allows for flexible configuration of signal routing, gain adjustments, and mixing parameters, enabling precise control over the audio output. This module can handle complex mixing tasks, such as applying different gain levels, phase adjustments, or filtering to individual channels before combining them into a final output signal. The platform ensures that the processed audio signals maintain high fidelity and low latency, making it suitable for real-time applications. The matrix mixing module may also include additional signal processing features, such as equalization, dynamic range compression, or spatial audio effects, to further enhance the audio output. By integrating these capabilities, the platform provides a versatile solution for professional audio applications requiring advanced signal routing and mixing.
4. The platform of claim 1 , further comprising: a system configuration component configured to provide pre-selected audio processing parameters to the digital signal processing component for at least one of the near-end audio devices, the digital signal processing component being further configured to apply the pre-selected parameters to the corresponding near-end input audio signal.
This invention relates to a platform for managing audio processing in communication systems, particularly for optimizing near-end audio signals in real-time applications. The problem addressed is the need for efficient and adaptive audio processing to enhance communication quality, such as in teleconferencing or voice-over-IP systems, where near-end audio devices (e.g., microphones) require dynamic adjustments to improve clarity and reduce noise. The platform includes a digital signal processing (DSP) component that processes input audio signals from near-end devices. A system configuration component provides pre-selected audio processing parameters to the DSP, which then applies these parameters to the corresponding near-end input audio signals. These parameters may include settings for noise reduction, echo cancellation, gain control, or other audio enhancements. The configuration component ensures that the DSP can dynamically adjust processing based on predefined settings, improving audio quality without manual intervention. The platform may also include a user interface for configuring these parameters, allowing users or administrators to customize audio processing for different scenarios. The DSP component processes the audio signals in real-time, ensuring minimal latency while maintaining high-quality output. This system is particularly useful in environments where audio conditions vary, such as in conference rooms or remote work setups, where consistent audio performance is critical. The invention aims to provide a scalable and adaptable solution for optimizing near-end audio in communication systems.
5. The platform of claim 4 , wherein the system configuration component is further configured to identify device information associated with the at least one near-end audio device, and retrieve one or more pre-selected audio processing parameters from a memory for said near-end audio device based on the identified device information.
This invention relates to audio processing systems, specifically for optimizing audio device performance based on device-specific configurations. The problem addressed is the need for automated, device-specific audio parameter adjustments to enhance audio quality without manual intervention. The system includes a platform with a system configuration component that identifies device information for at least one near-end audio device, such as a microphone or speaker. This component retrieves pre-selected audio processing parameters from a memory based on the identified device information. The parameters may include settings for noise reduction, equalization, or other audio enhancements tailored to the specific device's capabilities and characteristics. The system dynamically applies these parameters to improve audio output or input quality, ensuring optimal performance without requiring user adjustments. The platform may also include a user interface for managing these configurations, allowing users to modify or override the pre-selected parameters if needed. The system ensures compatibility and performance across different audio devices by leveraging stored device-specific profiles, reducing the need for manual tuning. This approach enhances user experience by automating audio optimization based on device-specific data.
6. The platform of claim 1 , wherein the digital signal processing component further comprises: a decryption module configured to decrypt one or more of the input audio signals, and an encryption module configured to encrypt one or more of the audio output signals.
This invention relates to a digital signal processing platform designed for secure audio signal handling. The platform processes input audio signals, which may include encrypted data, and generates audio output signals. A key feature is the inclusion of a decryption module that decrypts one or more input audio signals before processing, ensuring secure handling of sensitive audio data. Additionally, an encryption module encrypts one or more of the processed audio output signals, providing secure transmission or storage of the processed audio. The platform may also include other components such as analog-to-digital converters, digital-to-analog converters, and audio processing algorithms to enhance, filter, or analyze the audio signals. The encryption and decryption modules ensure that audio data remains protected throughout the processing pipeline, addressing security concerns in applications where audio signals contain confidential or proprietary information. This design is particularly useful in secure communication systems, medical audio processing, and other domains requiring robust data protection.
7. The platform of claim 1 , wherein the network audio library is configured to interface with the one or more near-end audio devices using a network audio control interface, and interface with the digital signal processing component using an audio interface.
This invention relates to a platform for managing audio processing in a networked environment. The problem addressed is the need for efficient and flexible audio signal routing and processing across multiple devices and digital signal processing (DSP) components. The platform includes a network audio library that serves as a central hub for audio data, enabling seamless communication between near-end audio devices (such as microphones, speakers, or other input/output devices) and DSP components responsible for processing the audio signals. The network audio library interfaces with near-end audio devices using a network audio control interface, which facilitates the transmission and reception of audio data over a network. This interface ensures low-latency and high-quality audio streaming between devices. Additionally, the library interfaces with the DSP component using an audio interface, allowing the processed audio signals to be further manipulated or optimized before being output to the near-end devices. The DSP component may perform tasks such as noise reduction, echo cancellation, or audio enhancement. By integrating these interfaces, the platform enables dynamic routing of audio signals between devices and processing components, improving the overall audio experience in applications such as teleconferencing, multimedia streaming, or audio production. The system ensures compatibility and interoperability between different audio devices and processing modules, reducing the complexity of managing audio workflows in networked environments.
8. The platform of claim 1 , wherein the virtual audio device driver is configured to interface with the conferencing software module using a first application programming interface, and interface with the digital signal processing component using a second application programming interface.
This invention relates to a platform for managing audio processing in conferencing applications. The problem addressed is the need for efficient and flexible audio signal processing in virtual conferencing environments, where multiple software components must interact seamlessly to provide high-quality audio. The platform includes a virtual audio device driver that acts as an intermediary between conferencing software and a digital signal processing (DSP) component. The virtual audio device driver is configured to interface with the conferencing software using a first application programming interface (API), allowing the conferencing software to send and receive audio data. Simultaneously, the driver interfaces with the DSP component using a second API, enabling real-time audio processing tasks such as noise reduction, echo cancellation, or audio enhancement. This dual-API architecture ensures compatibility and efficient data flow between the conferencing software and the DSP component, improving audio quality and performance in virtual meetings. The platform may also include additional components, such as a user interface for configuring audio settings or a network interface for transmitting processed audio data to remote participants. The overall system enhances the reliability and flexibility of audio processing in conferencing applications.
9. The platform of claim 4 , further comprising: a controller module configured to interface with the system configuration component using a control interface, interface with the network audio library using a third application programming interface, and interface with the digital signal processing component using a fourth application programming interface.
This invention relates to a platform for managing and processing audio signals in a networked environment. The platform addresses the challenge of integrating multiple audio processing components, such as system configuration, digital signal processing (DSP), and network audio libraries, into a cohesive system. The platform includes a controller module that acts as a central interface, facilitating communication between these components. The controller module connects to a system configuration component via a control interface, allowing for dynamic adjustments to system settings. It also interfaces with a network audio library through a third application programming interface (API), enabling access to audio data and network-based audio functions. Additionally, the controller module interacts with a DSP component using a fourth API, ensuring real-time processing of audio signals. This modular design allows for flexible integration of different audio processing elements, improving system scalability and adaptability. The platform is particularly useful in applications requiring coordinated audio signal management, such as telecommunication systems, multimedia streaming, or audio conferencing.
10. The platform of claim 1 , further comprising: a licensing module configured to determine a number of channels available to the digital signal processing component for receiving the near-end input audio signals based on one or more licenses associated with the platform.
This invention relates to a digital signal processing platform designed to enhance audio communication by managing near-end input audio signals. The platform includes a digital signal processing component that processes these signals to improve audio quality, such as noise reduction or echo cancellation. A key challenge addressed is efficiently allocating processing resources, particularly the number of audio channels available for processing, to optimize performance while adhering to licensing constraints. The platform includes a licensing module that dynamically determines the number of channels the digital signal processing component can use based on one or more licenses associated with the platform. This ensures that the system operates within licensed limits while maximizing available resources. The licensing module evaluates the licenses to enforce restrictions, such as the maximum number of concurrent audio channels permitted, and adjusts the processing component accordingly. This approach prevents unauthorized use of additional channels beyond the licensed capacity, ensuring compliance with licensing agreements while maintaining optimal audio processing performance. The platform may also include other components, such as an audio input interface for capturing near-end audio signals and an audio output interface for delivering processed signals to a communication device. The digital signal processing component applies algorithms to enhance audio quality, such as noise suppression or beamforming, based on the available channels. The licensing module ensures that these operations remain within the licensed channel limits, providing a scalable and compliant solution for audio processing in communication systems.
11. The platform of claim 1 , wherein the virtual audio device driver includes a mute logic module configured to synchronize a mute status of a given audio source across all other audio sources in the platform.
This invention relates to a platform for managing audio sources in a computing environment, addressing the challenge of maintaining consistent mute status across multiple audio sources. The platform includes a virtual audio device driver that interfaces with multiple audio sources, such as applications or hardware devices, to control audio output. A key feature is a mute logic module within the driver, which ensures that muting one audio source automatically updates the mute status of all other audio sources in the system. This synchronization prevents discrepancies where some sources remain active while others are muted, improving user experience and system consistency. The platform may also include additional components, such as a virtual audio mixer or a configuration interface, to further manage audio routing and settings. The mute logic module operates by detecting mute commands from any audio source and propagating the status change to all connected sources, ensuring uniform behavior across the platform. This solution is particularly useful in environments where multiple audio streams must be managed cohesively, such as in multimedia applications or virtual conferencing systems.
12. The platform of claim 1 , wherein the digital signal processing component further includes a clock synchronization module configured to synchronize the received input audio signals to a single clock.
This invention relates to a digital signal processing platform designed to handle multiple input audio signals. The primary problem addressed is the need to process and synchronize audio signals from different sources, which often operate on independent clocks, leading to timing mismatches and synchronization issues. The platform includes a digital signal processing component that processes the input audio signals to enhance audio quality, reduce noise, or perform other audio processing tasks. A key feature is a clock synchronization module within the digital signal processing component. This module synchronizes the received input audio signals to a single clock, ensuring that all signals are time-aligned. This synchronization is critical for applications requiring precise timing, such as audio conferencing, live broadcasting, or multi-channel audio systems. The platform may also include additional components for signal conditioning, such as filters, amplifiers, or equalizers, to further improve audio quality. The synchronization module may use techniques like sample rate conversion, buffering, or phase-locked loops to align the signals. By synchronizing the signals to a single clock, the platform ensures coherent processing and output, eliminating timing discrepancies between different audio sources. This solution is particularly useful in environments where multiple audio devices or streams must be combined or processed together.
13. The platform of claim 1 , further comprising: a resource monitoring module configured to collect usage information for computing resources in use by the platform, generate one or more alerts based thereon, and provide said alerts to a user interface for presentation to a user.
A system for managing computing resources includes a platform that monitors and optimizes resource utilization. The platform collects usage data for computing resources such as processing power, memory, storage, and network bandwidth. A resource monitoring module within the platform analyzes this data to detect inefficiencies, overutilization, or potential failures. The module generates alerts when predefined thresholds are exceeded or when anomalies are detected. These alerts are displayed to users through a user interface, providing real-time visibility into resource performance. The alerts may include recommendations for corrective actions, such as scaling resources, reallocating workloads, or scheduling maintenance. The system ensures efficient resource allocation, prevents downtime, and enhances overall system performance by proactively identifying and addressing issues. The monitoring module may also log historical data for trend analysis and capacity planning. This approach helps organizations maintain optimal resource usage while minimizing costs and improving reliability.
14. A computer-implemented method of audio processing for a conferencing environment, comprising: receiving input audio signals at a plurality of audio sources, the audio sources comprising a virtual audio device driver and a network audio library, wherein the receiving comprises: receiving far-end input audio signals at the virtual audio device driver from a conferencing software module, and receiving near-end input audio signals at the network audio library from one or more near-end audio devices; and processing the input audio signals using a digital signal processing component, the processing comprising: applying acoustic echo cancellation techniques to one or more of the near-end input audio signals, and generating audio output signals based on the input audio signals, wherein each of the virtual audio device driver, the network audio library, and the digital signal processing component is stored on a local computing device connected to said one or more near-end audio devices comprising said conferencing software module, and wherein the virtual audio device driver appears as a hardware audio device to the conferencing software module.
This invention relates to audio processing in conferencing environments, addressing challenges in managing audio signals from multiple sources while minimizing echo and ensuring seamless integration with conferencing software. The system receives input audio signals from both far-end and near-end sources. Far-end signals are captured by a virtual audio device driver, which interfaces with conferencing software and appears as a hardware audio device to the software. Near-end signals are received by a network audio library from local audio devices like microphones. A digital signal processing component processes these signals, applying acoustic echo cancellation to near-end inputs to reduce feedback. The processed signals are then combined to generate output audio for the conferencing environment. All components—virtual audio device driver, network audio library, and digital signal processing—are stored locally on a computing device running the conferencing software, ensuring low-latency, efficient audio handling. The system improves audio quality in conferencing by dynamically managing multiple audio streams and mitigating echo artifacts.
15. The method of claim 14 , wherein the processing further comprises mixing two or more of the near-end input audio signals to generate an automix output signal.
This invention relates to audio signal processing, specifically automixing techniques for combining multiple audio inputs. The problem addressed is the need to automatically blend multiple audio sources, such as microphones in a conference or recording environment, while suppressing unwanted noise and feedback. The invention provides a method for processing near-end input audio signals, which are audio signals captured from a local environment, such as a conference room. The method includes analyzing the input signals to determine their characteristics, such as speech presence, noise levels, and feedback risk. Based on this analysis, the method dynamically adjusts the gain or attenuation of each input signal to optimize the output audio quality. The processing further includes mixing two or more of the near-end input audio signals to generate an automix output signal. This mixing step ensures that the combined output is clear and free from interference, while maintaining the integrity of the desired audio content. The method may also include additional steps such as noise suppression, echo cancellation, and feedback suppression to enhance the overall audio quality. The invention is particularly useful in applications where multiple audio sources need to be automatically managed, such as video conferencing, live broadcasting, and audio recording.
16. The method of claim 14 , wherein the generating comprises using a matrix mixer to generate the audio output signals.
This invention relates to audio signal processing, specifically methods for generating audio output signals from input signals. The problem addressed is the need for efficient and accurate audio signal mixing, particularly in systems requiring precise control over signal combinations. The method involves generating audio output signals from input signals, where the input signals are processed to produce the desired output. A key aspect is the use of a matrix mixer in the generation process. A matrix mixer is a device or algorithm that combines multiple input audio signals into one or more output signals using a matrix of coefficients. This allows for flexible and precise control over how input signals contribute to the final output, enabling applications such as spatial audio rendering, beamforming, or multi-channel audio mixing. The matrix mixer applies a set of coefficients to the input signals, which may be adjusted dynamically to achieve specific audio effects or optimize signal quality. The coefficients can be derived from predefined settings, user inputs, or adaptive algorithms that respond to real-time conditions. The resulting audio output signals are then provided for further processing or direct playback. This approach improves upon traditional mixing techniques by offering greater flexibility and precision in signal combination, making it suitable for advanced audio systems where dynamic and accurate mixing is required.
17. The method of claim 14 , further comprising: providing pre-selected audio processing parameters to the digital signal processing component for at least one of the near-end audio devices, wherein the processing further comprises applying the pre-selected parameters to the corresponding near-end input audio signal.
This invention relates to audio processing systems, specifically methods for enhancing audio signals in communication devices. The problem addressed is the need for improved audio quality in real-time communication, particularly when multiple near-end audio devices (such as microphones) are involved. The invention provides a method for processing audio signals from these devices using a digital signal processing (DSP) component. The DSP component receives input audio signals from the near-end devices and applies processing techniques to enhance the audio quality. The processing may include noise reduction, echo cancellation, or other audio enhancements. Additionally, the method includes providing pre-selected audio processing parameters to the DSP component for at least one of the near-end audio devices. These parameters are applied to the corresponding input audio signal to further refine the audio output. The pre-selected parameters may be tailored to specific audio conditions or device characteristics, ensuring optimized performance. This approach allows for dynamic and adaptive audio processing, improving clarity and reducing interference in communication systems. The invention is particularly useful in environments where multiple audio sources are present, such as conference calls or multi-microphone setups.
18. The method of claim 17 , further comprising: identifying device information associated with the at least one near-end audio device; and retrieving one or more pre-selected audio processing parameters from a memory for said near-end audio device based on the identified device information.
This invention relates to audio processing systems, specifically methods for dynamically adjusting audio processing parameters based on near-end device characteristics. The problem addressed is the need to optimize audio quality in communication systems by tailoring processing parameters to the specific capabilities and configurations of near-end audio devices, such as microphones or speakers, to improve clarity and performance. The method involves identifying device information for at least one near-end audio device, which may include hardware specifications, firmware versions, or configuration settings. This information is used to retrieve pre-selected audio processing parameters from a memory. The parameters are pre-configured for the specific device to ensure optimal performance, such as noise suppression levels, equalization settings, or gain adjustments. By dynamically applying these parameters, the system adapts to different devices without manual intervention, enhancing audio quality across various hardware configurations. The approach improves consistency and reduces setup complexity in multi-device environments.
19. The method of claim 14 , wherein the processing further comprises: decrypting one or more of the input audio signals; and encrypting one or more of the audio output signals.
This invention relates to audio signal processing systems, specifically methods for handling encrypted audio signals in real-time applications. The problem addressed is the secure transmission and processing of audio data while maintaining confidentiality and integrity. The method involves receiving multiple input audio signals, which may be encrypted, and processing them to generate one or more audio output signals. The processing step includes decrypting the input signals if they are encrypted, performing necessary audio processing (such as mixing, filtering, or beamforming), and then encrypting the resulting output signals before transmission. This ensures that audio data remains protected throughout the processing pipeline, preventing unauthorized access or tampering. The system is designed for applications where secure audio communication is critical, such as military, government, or enterprise environments. The encryption and decryption steps are integrated into the processing workflow, allowing seamless handling of encrypted audio without requiring separate decryption and encryption modules. The method supports real-time operation, ensuring low-latency processing while maintaining security.
20. The method of claim 14 , further comprising: determining a number of channels available to the digital signal processing component for receiving the near-end input audio signals based on one or more licenses associated with the conferencing environment.
This invention relates to digital signal processing in conferencing environments, specifically addressing the challenge of dynamically managing audio channels based on licensing constraints. The method involves a digital signal processing component that processes near-end input audio signals from multiple participants in a conferencing system. The component filters these signals to remove echo and noise, then applies beamforming to enhance speech clarity. The method further includes determining the number of available channels for receiving near-end input audio signals based on one or more licenses associated with the conferencing environment. This ensures that the system operates within licensed capacity limits while optimizing audio quality. The licensing information may be stored locally or retrieved from a remote server, and the system dynamically adjusts channel allocation accordingly. This approach prevents unauthorized use of additional channels beyond the licensed capacity while maintaining efficient audio processing. The method is particularly useful in large-scale conferencing systems where licensing compliance and audio quality are critical.
21. The method of claim 14 , further comprising: synchronizing the received input audio signals to a single clock.
This invention relates to audio signal processing, specifically methods for handling multiple input audio signals. The problem addressed is the misalignment of audio signals from different sources, which can cause synchronization issues in applications like audio conferencing, live sound mixing, or multi-microphone recording. The invention provides a solution by synchronizing received input audio signals to a single clock, ensuring all signals are time-aligned. The method involves receiving multiple input audio signals from different sources, such as microphones or audio devices. These signals may arrive at different times or with varying delays due to differences in processing, transmission, or physical distance. To resolve this, the method synchronizes all input signals to a single reference clock. This synchronization ensures that the audio signals are aligned in time, preventing phase differences or timing errors that could degrade audio quality or intelligibility. The synchronization process may involve adjusting the timing of each input signal to match the reference clock, compensating for inherent delays in each signal path. This can be achieved through techniques such as buffering, time-stamping, or digital signal processing. The synchronized signals can then be processed further, such as mixing, filtering, or transmitting, with improved coherence and clarity. This method is particularly useful in applications requiring precise timing, such as real-time audio conferencing, live event sound reinforcement, or multi-channel audio recording. By ensuring all audio signals are aligned, the invention enhances audio quality and reduces artifacts caused by misalignment.
22. The method of claim 14 , further comprising: collecting usage information for computing resources in use by the conferencing environment; generating one or more alerts based thereon; and providing said alerts to a user interface for presentation to a user.
This invention relates to monitoring and managing computing resources in a conferencing environment, such as a virtual meeting or collaboration platform. The problem addressed is the lack of real-time visibility into resource utilization, which can lead to inefficiencies, performance degradation, or unexpected costs. The invention provides a solution by collecting detailed usage information for computing resources consumed by the conferencing environment, such as processing power, memory, network bandwidth, or storage. This data is analyzed to detect anomalies, bottlenecks, or excessive consumption, and alerts are generated when predefined thresholds are exceeded or when abnormal patterns are detected. The alerts are then presented to a user interface, allowing administrators or users to take corrective actions, such as optimizing resource allocation, scaling infrastructure, or troubleshooting issues. The system may also support historical tracking and reporting to identify long-term trends or recurring problems. By providing real-time insights and automated alerts, the invention helps ensure efficient resource utilization, improved performance, and cost savings in conferencing environments.
23. The method of claim 14 , further comprising: synchronizing a mute status of a given audio source across all other audio sources in the conferencing environment.
This invention relates to audio conferencing systems and addresses the problem of inconsistent mute statuses across multiple audio sources in a conferencing environment. The method involves synchronizing the mute status of a given audio source across all other audio sources in the system. When one audio source is muted or unmuted, the mute status is automatically updated for all other audio sources in the environment. This ensures that participants in the conference experience a consistent audio experience, preventing unintended audio disruptions or overlaps. The synchronization process may involve real-time communication between the audio sources or a central control system that manages the mute status for all connected devices. The method may also include detecting changes in mute status, such as when a participant manually mutes or unmutes their microphone, and propagating those changes to all other audio sources. This approach improves collaboration by maintaining uniform audio control across the conferencing environment.
24. A conferencing system, comprising: one or more near-end audio devices configured to capture near-end audio signals; and a local computing device connected to said one or more near-end audio devices and comprising one or more processors and one or more memory devices, the one or more memory devices storing a plurality of computer software programs configured to be executed by the one or more processors, the programs comprising: a conferencing software module configured to receive far-end audio signals from at least one remote server communicatively coupled to the local computing device; a virtual audio device driver configured to receive the far-end audio signals from the conferencing software module; a network audio library configured to receive the near-end audio signals from the one or more near-end audio devices; and a digital signal processing component configured to receive the near-end audio signals from the network audio library, receive the far-end audio signals from the virtual audio device driver, and generate audio output signals based on the received signals, wherein the digital signal processing component comprises an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end audio signals, and wherein the virtual audio device driver appears as a hardware audio device to the conferencing software module.
This conferencing system addresses the challenge of managing audio signals in real-time communication environments, particularly in reducing echo and improving audio clarity during conferences. The system includes near-end audio devices, such as microphones, that capture audio signals from participants in a local environment. These signals are processed by a local computing device equipped with one or more processors and memory devices storing multiple software programs. The conferencing software module receives far-end audio signals from remote servers, representing audio from distant participants. A virtual audio device driver interfaces with the conferencing software, acting as a hardware audio device to ensure compatibility and seamless integration. The network audio library captures near-end audio signals from the local microphones. A digital signal processing component processes both near-end and far-end audio signals, applying acoustic echo cancellation to the near-end signals to eliminate feedback and enhance audio quality. The system generates optimized audio output signals for playback, ensuring clear communication without echo interference. The virtual audio device driver's hardware-like appearance simplifies integration with existing conferencing software, while the digital signal processing component dynamically adjusts audio signals for optimal performance. This architecture improves real-time audio conferencing by minimizing echo and maintaining high-fidelity audio transmission.
25. The conferencing system of claim 24 , wherein the digital signal processing component further comprises an automixing module configured to mix two or more of the near-end audio signals to generate an automix output signal.
This invention relates to conferencing systems designed to improve audio communication in multi-participant environments. The system addresses the challenge of managing multiple audio signals from different participants, ensuring clear and intelligible audio output while minimizing background noise and interference. The conferencing system includes a digital signal processing (DSP) component that processes near-end audio signals, which are audio inputs from participants in the same physical location. The DSP component enhances these signals by applying noise reduction, echo cancellation, and other audio processing techniques to improve clarity. A key feature of the system is an automixing module within the DSP component. This module dynamically combines two or more near-end audio signals into a single automix output signal. The automixing process ensures that only the most relevant audio sources are prioritized, reducing clutter and improving the listening experience for remote participants. The automixing module may use adaptive algorithms to adjust mixing parameters based on factors such as signal strength, speech activity, or user preferences, ensuring optimal audio quality in real-time. By integrating automixing with other audio processing functions, the system provides a seamless and efficient solution for managing complex audio environments in conferencing applications. This enhances collaboration by delivering clearer, more focused audio output.
26. The conferencing system of claim 24 , wherein the digital signal processing component further comprises a matrix mixing module configured to generate the audio output signals.
This invention relates to conferencing systems designed to improve audio communication in multi-party environments. The system addresses the challenge of managing and optimizing audio signals from multiple participants in real-time to enhance clarity and reduce interference. The conferencing system includes a digital signal processing component that processes audio inputs from various sources, such as microphones or remote participants, to produce high-quality audio output signals. A key feature is a matrix mixing module within the digital signal processing component. This module dynamically adjusts and combines the audio signals from different participants based on factors like signal strength, participant activity, or user preferences. The matrix mixing module ensures that the audio output signals are balanced, minimizing background noise and prioritizing active speakers. The system may also include additional components, such as noise suppression modules or echo cancellation modules, to further refine the audio quality. The overall goal is to provide a seamless and intelligible audio experience for all participants in a conferencing environment.
27. The conferencing system of claim 24 , wherein the programs further comprise a system configuration component configured to provide pre-selected audio processing parameters to the digital signal processing component for at least one of the near-end audio devices, the digital signal processing component being further configured to apply the pre-selected parameters to the corresponding near-end audio signal.
This invention relates to a conferencing system designed to enhance audio processing for near-end participants. The system addresses the challenge of optimizing audio quality in conferencing environments by dynamically adjusting audio processing parameters based on pre-selected configurations. The conferencing system includes multiple near-end audio devices, such as microphones and speakers, and a digital signal processing (DSP) component that processes audio signals from these devices. A system configuration component provides pre-selected audio processing parameters to the DSP component, which then applies these parameters to the corresponding near-end audio signals. This allows for tailored audio processing, such as noise reduction, echo cancellation, or gain control, to be applied based on predefined settings. The system ensures consistent and high-quality audio performance by leveraging pre-configured parameters, reducing the need for real-time adjustments and improving the overall conferencing experience. The invention is particularly useful in environments where specific audio processing requirements must be met for different participants or devices.
28. The conferencing system of claim 27 , wherein the system configuration component is further configured to identify device information associated with the at least one near-end audio device, and retrieve one or more pre-selected audio processing parameters from the at least one memory for said near-end audio device based on the identified device information.
This invention relates to conferencing systems designed to optimize audio processing for near-end devices. The system addresses the challenge of ensuring high-quality audio in conferencing environments by dynamically adjusting audio processing parameters based on the specific characteristics of the near-end audio devices involved. The system includes a configuration component that identifies device information for each near-end audio device, such as its type, model, or capabilities. Using this information, the system retrieves pre-selected audio processing parameters from memory that are tailored to the identified device. These parameters may include settings for noise reduction, echo cancellation, gain control, or other audio enhancements. By automatically applying device-specific parameters, the system ensures consistent and optimized audio performance across different near-end devices in a conferencing session. This approach eliminates the need for manual configuration and improves the overall user experience by adapting to the unique requirements of each device. The system may also include components for managing audio streams, processing audio signals, and interfacing with far-end devices to facilitate real-time communication. The invention aims to enhance audio quality in conferencing applications by leveraging device-specific optimizations.
29. The conferencing system of claim 24 , wherein the digital signal processing component further comprises: a decryption module configured to decrypt one or more of the input audio signals, and an encryption module configured to encrypt one or more of the audio output signals.
This invention relates to a conferencing system designed to enhance secure communication in multi-party audio conferencing environments. The system addresses the challenge of maintaining secure and efficient audio signal processing in real-time conferencing applications, where participants may need encrypted communication to protect sensitive information. The conferencing system includes a digital signal processing (DSP) component that processes input audio signals from multiple participants. The DSP component further includes a decryption module and an encryption module. The decryption module decrypts one or more of the input audio signals received from participants, ensuring that encrypted audio data is properly decoded before further processing. The encryption module encrypts one or more of the audio output signals before transmission to participants, ensuring that the audio data remains secure during transmission. This dual encryption-decryption functionality allows the system to handle both incoming and outgoing audio streams securely, preventing unauthorized access to the audio content. The conferencing system may also include additional components such as a noise reduction module, an echo cancellation module, and a beamforming module, which collectively improve audio quality by reducing background noise, eliminating echo, and focusing on the primary speaker. The DSP component processes the audio signals in real-time, ensuring minimal latency while maintaining high-quality audio output. The system is particularly useful in enterprise, government, or healthcare settings where secure and clear communication is critical.
30. The conferencing system of claim 29 , wherein at least one of the near-end audio devices is configured to encrypt the near-end audio signals prior to transmitting the signals to the network audio library.
This invention relates to a conferencing system designed to enhance audio communication in multi-party conferencing environments. The system addresses the challenge of managing and processing audio signals from multiple participants in real-time, ensuring clarity, synchronization, and security. The conferencing system includes a network audio library that receives and processes audio signals from near-end audio devices, such as microphones or other input devices, and far-end audio devices, such as speakers or other output devices. The system dynamically adjusts audio processing parameters, such as gain, noise reduction, and echo cancellation, based on the characteristics of the audio signals and the network conditions. Additionally, the system supports encryption of near-end audio signals before transmission to the network audio library, ensuring secure communication. The system may also include a synchronization module to align audio signals from different participants, improving the overall conferencing experience. The conferencing system is designed to operate in various network environments, including low-bandwidth or high-latency conditions, while maintaining high-quality audio output.
31. The conferencing system of claim 24 , wherein the one or more near-end audio devices include a conferencing device comprising at least one microphone, the conferencing device being connected to the computing device using a local network connection.
This invention relates to conferencing systems designed to improve audio communication in multi-party environments. The system addresses the challenge of managing audio inputs from multiple near-end devices, such as microphones, to enhance clarity and reduce interference during conferences. The conferencing system includes a computing device that processes audio signals from one or more near-end audio devices, which may include a dedicated conferencing device with at least one microphone. This conferencing device connects to the computing device via a local network, enabling seamless integration and centralized audio management. The system dynamically adjusts audio processing parameters based on the number and type of near-end devices, ensuring optimal sound quality and minimizing background noise. By leveraging network connectivity, the system allows for flexible deployment and scalability, accommodating various conferencing setups. The invention aims to provide a robust solution for enhancing audio performance in collaborative environments, particularly in scenarios where multiple participants contribute to the conversation.
32. The conferencing system of claim 31 , wherein the conferencing device further comprises at least one speaker for playing the far-end audio signals.
This invention relates to conferencing systems designed to improve audio communication in multi-party interactions. The system addresses the challenge of delivering clear and synchronized audio from remote participants (far-end audio signals) to local participants in a conference setting. The conferencing device includes at least one speaker specifically configured to play these far-end audio signals, ensuring that participants can hear remote voices distinctly. The system likely integrates with other components, such as microphones, processing units, and network interfaces, to capture, transmit, and reproduce audio signals efficiently. The speaker may be optimized for directional sound projection, noise cancellation, or adaptive volume control to enhance clarity in various environments. The invention aims to provide a seamless audio experience by minimizing latency, distortion, or interference, thereby improving collaboration in virtual or hybrid meetings. The conferencing device may also include features like echo cancellation, beamforming, or spatial audio processing to further refine sound quality. The system is particularly useful in professional, educational, or personal conferencing applications where reliable audio transmission is critical.
33. The conferencing system of claim 24 , wherein the programs further comprise a licensing module configured to determine a number of channels available to the digital signal processing component for receiving the near-end audio signals based on one or more licenses associated with the system.
This invention relates to a conferencing system that processes audio signals in real-time to enhance communication quality. The system addresses the challenge of managing multiple audio channels efficiently, particularly in environments where near-end audio signals from different participants need to be processed simultaneously. The system includes a digital signal processing (DSP) component that receives and processes these near-end audio signals to improve clarity, reduce noise, and optimize audio mixing for far-end participants. A key feature is a licensing module that dynamically determines the number of available channels for the DSP component based on one or more licenses associated with the system. This ensures that the system operates within licensed capacity limits while maximizing the number of active audio channels for optimal conferencing performance. The licensing module may enforce restrictions based on the type or tier of the license, allowing scalable deployment across different user needs. The system may also include additional modules for audio enhancement, such as echo cancellation, noise suppression, and beamforming, to further improve audio quality. The licensing module ensures compliance with licensing terms while maintaining high-quality audio processing for all participants.
34. The conferencing system of claim 24 , wherein the virtual audio device driver comprises a mute logic module configured to synchronize, across the system, a mute status associated with at least one of the one or more near-end devices and the conferencing software module.
A conferencing system includes a virtual audio device driver that manages audio input and output for multiple near-end devices in a conferencing session. The system addresses the challenge of maintaining consistent audio control across different devices and software components during a conference call. The virtual audio device driver acts as an intermediary between the conferencing software and the near-end devices, ensuring seamless audio routing and synchronization. The virtual audio device driver includes a mute logic module that synchronizes mute status across the system. This module ensures that the mute state of any near-end device is reflected in the conferencing software and vice versa. For example, if a user mutes their microphone on a near-end device, the mute status is automatically updated in the conferencing software, preventing audio feedback or unintended transmission. Similarly, if the conferencing software mutes a participant, the mute status is propagated to the corresponding near-end device. This synchronization prevents discrepancies in mute states, improving user experience and call quality. The system may also include additional features such as audio mixing, device selection, and real-time audio processing to enhance conferencing functionality.
35. The conferencing system of claim 24 , wherein the digital signal processing component further comprises a clock synchronization module configured to synchronize the near-end and far-end audio signals to a single clock.
This invention relates to conferencing systems designed to improve audio synchronization between participants in a communication session. The system addresses the problem of audio delay and misalignment in multi-party conferences, where differences in network latency and processing times can cause far-end audio signals to arrive out of sync with near-end audio, leading to disruptions in natural conversation flow. The conferencing system includes a digital signal processing component that processes audio signals from multiple participants. A key feature is a clock synchronization module within this component, which synchronizes near-end and far-end audio signals to a single clock. This ensures that audio from different sources is aligned in time, reducing or eliminating the perception of delay and improving the overall conferencing experience. The synchronization process compensates for variations in network transmission times and processing delays, ensuring that all audio streams are time-aligned at the receiving end. The system may also include other components, such as audio capture and playback modules, which handle the acquisition and output of audio signals. The digital signal processing component further processes these signals to enhance clarity, suppress background noise, and optimize audio quality. By synchronizing all audio streams to a unified clock, the system ensures that participants hear each other's contributions in real time, facilitating smoother and more natural interactions. This synchronization is particularly valuable in professional settings, such as virtual meetings or collaborative workspaces, where precise timing is critical.
36. The conferencing system of claim 24 , further comprising a user interface, wherein the plurality of programs further comprise a resource monitoring module configured to collect usage information for computing resources in use by the programs, generate one or more alerts based thereon, and provide said alerts to the user interface for presentation to a user.
A conferencing system includes a plurality of programs that enable real-time communication between participants, such as video conferencing, file sharing, and messaging. The system monitors computing resources used by these programs, including CPU, memory, and network bandwidth, to ensure efficient operation. A resource monitoring module collects usage data, analyzes it to detect potential issues like resource exhaustion or performance bottlenecks, and generates alerts when thresholds are exceeded. These alerts are displayed to users via a user interface, allowing them to take corrective actions, such as adjusting settings or terminating resource-intensive processes. The system may also prioritize critical functions to maintain stable conferencing performance. This approach helps prevent disruptions during meetings by proactively managing system resources and providing users with visibility into resource consumption. The resource monitoring module operates in the background, ensuring minimal impact on the conferencing experience while maintaining system stability.
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May 28, 2019
March 15, 2022
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